Merge branch 'v3.1-rc10' into for-3.2/core

Conflicts:
	block/blk-core.c
	include/linux/blkdev.h

Signed-off-by: Jens Axboe <axboe@kernel.dk>

16 files changed, 188 insertions, 106 deletions

diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 50abf5b..8816804 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -1,5 +1,10 @@
 # ALSA PCI drivers
 
+config SND_TEA575X
+	tristate
+	depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
+	default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
+
 menuconfig SND_PCI
 	bool "PCI sound devices"
 	depends on PCI
@@ -563,11 +568,6 @@ config SND_FM801_TEA575X_BOOL
 	  FM801 chip with a TEA5757 tuner (MediaForte SF256-PCS, SF256-PCP and
 	  SF64-PCR) into the snd-fm801 driver.
 
-config SND_TEA575X
-	tristate
-	depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
-	default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
-
 source "sound/pci/hda/Kconfig"
 
 config SND_HDSP
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 200c9a1..a872d0a 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1909,6 +1909,7 @@ static unsigned int ad1981_jacks_whitelist[] = {
 	0x103c0944, /* HP nc6220 */
 	0x103c0934, /* HP nc8220 */
 	0x103c006d, /* HP nx9105 */
+	0x103c300d, /* HP Compaq dc5100 SFF(PT003AW) */
 	0x17340088, /* FSC Scenic-W */
 	0 /* end */
 };
diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index 65b7ca1..bd47521 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -631,13 +631,12 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32 control_count,
 	if (!p_cache)
 		return NULL;
 
-	p_cache->p_info =
-		kmalloc(sizeof(*p_cache->p_info) * control_count, GFP_KERNEL);
+	p_cache->p_info = kzalloc(sizeof(*p_cache->p_info) * control_count,
+				  GFP_KERNEL);
 	if (!p_cache->p_info) {
 		kfree(p_cache);
 		return NULL;
 	}
-	memset(p_cache->p_info, 0, sizeof(*p_cache->p_info) * control_count);
 	p_cache->cache_size_in_bytes = size_in_bytes;
 	p_cache->control_count = control_count;
 	p_cache->p_cache = p_dsp_control_buffer;
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index e4d76a2..579fc0dce 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2625,16 +2625,19 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 	int err;
 
 	snd_azf3328_dbgcallenter();
-	if (dev >= SNDRV_CARDS)
-		return -ENODEV;
+	if (dev >= SNDRV_CARDS) {
+		err = -ENODEV;
+		goto out;
+	}
 	if (!enable[dev]) {
 		dev++;
-		return -ENOENT;
+		err = -ENOENT;
+		goto out;
 	}
 
 	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
 	if (err < 0)
-		return err;
+		goto out;
 
 	strcpy(card->driver, "AZF3328");
 	strcpy(card->shortname, "Aztech AZF3328 (PCI168)");
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index f9123f0..32b02d9 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -68,6 +68,7 @@ MODULE_PARM_DESC(enable, "Enable FM801 soundcard.");
 module_param_array(tea575x_tuner, int, NULL, 0444);
 MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only).");
 
+#define TUNER_DISABLED		(1<<3)
 #define TUNER_ONLY		(1<<4)
 #define TUNER_TYPE_MASK		(~TUNER_ONLY & 0xFFFF)
 
@@ -1150,7 +1151,8 @@ static int snd_fm801_free(struct fm801 *chip)
 
       __end_hw:
 #ifdef CONFIG_SND_FM801_TEA575X_BOOL
-	snd_tea575x_exit(&chip->tea);
+	if (!(chip->tea575x_tuner & TUNER_DISABLED))
+		snd_tea575x_exit(&chip->tea);
 #endif
 	if (chip->irq >= 0)
 		free_irq(chip->irq, chip);
@@ -1236,7 +1238,6 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 	    (tea575x_tuner & TUNER_TYPE_MASK) < 4) {
 		if (snd_tea575x_init(&chip->tea)) {
 			snd_printk(KERN_ERR "TEA575x radio not found\n");
-			snd_fm801_free(chip);
 			return -ENODEV;
 		}
 	} else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) {
@@ -1251,11 +1252,15 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 		}
 		if (tea575x_tuner == 4) {
 			snd_printk(KERN_ERR "TEA575x radio not found\n");
-			snd_fm801_free(chip);
-			return -ENODEV;
+			chip->tea575x_tuner = TUNER_DISABLED;
 		}
 	}
-	strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card));
+	if (!(chip->tea575x_tuner & TUNER_DISABLED)) {
+		strlcpy(chip->tea.card,
+			snd_fm801_tea575x_gpios[(tea575x_tuner &
+						 TUNER_TYPE_MASK) - 1].name,
+			sizeof(chip->tea.card));
+	}
 #endif
 
 	*rchip = chip;
diff --git a/sound/pci/hda/alc268_quirks.c b/sound/pci/hda/alc268_quirks.c
index be58bf2..2e5876c 100644
--- a/sound/pci/hda/alc268_quirks.c
+++ b/sound/pci/hda/alc268_quirks.c
@@ -476,8 +476,8 @@ static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
 
 static const struct alc_config_preset alc268_presets[] = {
 	[ALC267_QUANTA_IL1] = {
-		.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
-			    alc268_capture_nosrc_mixer },
+		.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_nosrc_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc267_quanta_il1_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -492,8 +492,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_3ST] = {
-		.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
+		.mixers = { alc268_base_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_alt_mixer,
 		.init_verbs = { alc268_base_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
 		.dac_nids = alc268_dac_nids,
@@ -507,8 +507,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.input_mux = &alc268_capture_source,
 	},
 	[ALC268_TOSHIBA] = {
-		.mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
+		.mixers = { alc268_toshiba_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_alt_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_toshiba_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -525,8 +525,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ACER] = {
-		.mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
+		.mixers = { alc268_acer_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_alt_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -543,8 +543,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ACER_DMIC] = {
-		.mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
+		.mixers = { alc268_acer_dmic_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_alt_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -561,9 +561,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ACER_ASPIRE_ONE] = {
-		.mixers = { alc268_acer_aspire_one_mixer,
-			    alc268_beep_mixer,
-			    alc268_capture_nosrc_mixer },
+		.mixers = { alc268_acer_aspire_one_mixer, alc268_beep_mixer},
+		.cap_mixer = alc268_capture_nosrc_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_aspire_one_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -579,8 +578,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_DELL] = {
-		.mixers = { alc268_dell_mixer, alc268_beep_mixer,
-			    alc268_capture_nosrc_mixer },
+		.mixers = { alc268_dell_mixer, alc268_beep_mixer},
+		.cap_mixer = alc268_capture_nosrc_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_dell_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -596,8 +595,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ZEPTO] = {
-		.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
+		.mixers = { alc268_base_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_alt_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_toshiba_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -616,7 +615,8 @@ static const struct alc_config_preset alc268_presets[] = {
 	},
 #ifdef CONFIG_SND_DEBUG
 	[ALC268_TEST] = {
-		.mixers = { alc268_test_mixer, alc268_capture_mixer },
+		.mixers = { alc268_test_mixer },
+		.cap_mixer = alc268_capture_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_volume_init_verbs,
 				alc268_beep_init_verbs },
diff --git a/sound/pci/hda/alc269_quirks.c b/sound/pci/hda/alc269_quirks.c
index 14fdcf2..5ac0e21 100644
--- a/sound/pci/hda/alc269_quirks.c
+++ b/sound/pci/hda/alc269_quirks.c
@@ -531,17 +531,10 @@ static const struct snd_pci_quirk alc269_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
-		      ALC269_DMIC),
-	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
-		      ALC269_DMIC),
-	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC),
-	SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC),
 	SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO),
 	SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
 	SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC),
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3e7850c..f3aefef 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -579,9 +579,13 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
 		return -1;
 	}
 	recursive++;
-	for (i = 0; i < nums; i++)
+	for (i = 0; i < nums; i++) {
+		unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i]));
+		if (type == AC_WID_PIN || type == AC_WID_AUD_OUT)
+			continue;
 		if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0)
 			return i;
+	}
 	return -1;
 }
 EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 28ce17d..c34f730 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -144,25 +144,17 @@ static int cea_sampling_frequencies[8] = {
 	SNDRV_PCM_RATE_192000,	/* 7: 192000Hz */
 };
 
-static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid,
+static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid,
 					int byte_index)
 {
 	unsigned int val;
 
 	val = snd_hda_codec_read(codec, nid, 0,
 					AC_VERB_GET_HDMI_ELDD, byte_index);
-
 #ifdef BE_PARANOID
 	printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val);
 #endif
-
-	if ((val & AC_ELDD_ELD_VALID) == 0) {
-		snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n",
-								byte_index);
-		val = 0;
-	}
-
-	return val & AC_ELDD_ELD_DATA;
+	return val;
 }
 
 #define GRAB_BITS(buf, byte, lowbit, bits) 		\
@@ -344,11 +336,26 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
 	if (!buf)
 		return -ENOMEM;
 
-	for (i = 0; i < size; i++)
-		buf[i] = hdmi_get_eld_byte(codec, nid, i);
+	for (i = 0; i < size; i++) {
+		unsigned int val = hdmi_get_eld_data(codec, nid, i);
+		if (!(val & AC_ELDD_ELD_VALID)) {
+			if (!i) {
+				snd_printd(KERN_INFO
+					   "HDMI: invalid ELD data\n");
+				ret = -EINVAL;
+				goto error;
+			}
+			snd_printd(KERN_INFO
+				  "HDMI: invalid ELD data byte %d\n", i);
+			val = 0;
+		} else
+			val &= AC_ELDD_ELD_DATA;
+		buf[i] = val;
+	}
 
 	ret = hdmi_update_eld(eld, buf, size);
 
+error:
 	kfree(buf);
 	return ret;
 }
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index be69822..e9a2a87 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1924,7 +1924,8 @@ static unsigned int azx_via_get_position(struct azx *chip,
 }
 
 static unsigned int azx_get_position(struct azx *chip,
-				     struct azx_dev *azx_dev)
+				     struct azx_dev *azx_dev,
+				     bool with_check)
 {
 	unsigned int pos;
 	int stream = azx_dev->substream->stream;
@@ -1940,7 +1941,7 @@ static unsigned int azx_get_position(struct azx *chip,
 	default:
 		/* use the position buffer */
 		pos = le32_to_cpu(*azx_dev->posbuf);
-		if (chip->position_fix[stream] == POS_FIX_AUTO) {
+		if (with_check && chip->position_fix[stream] == POS_FIX_AUTO) {
 			if (!pos || pos == (u32)-1) {
 				printk(KERN_WARNING
 				       "hda-intel: Invalid position buffer, "
@@ -1964,7 +1965,7 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
 	struct azx *chip = apcm->chip;
 	struct azx_dev *azx_dev = get_azx_dev(substream);
 	return bytes_to_frames(substream->runtime,
-			       azx_get_position(chip, azx_dev));
+			       azx_get_position(chip, azx_dev, false));
 }
 
 /*
@@ -1987,7 +1988,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
 		return -1;	/* bogus (too early) interrupt */
 
 	stream = azx_dev->substream->stream;
-	pos = azx_get_position(chip, azx_dev);
+	pos = azx_get_position(chip, azx_dev, true);
 
 	if (WARN_ONCE(!azx_dev->period_bytes,
 		      "hda-intel: zero azx_dev->period_bytes"))
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 47d6ffc..c45f3e6 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -375,7 +375,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
 static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
 			 unsigned int *idxp)
 {
-	int i;
+	int i, idx;
 	hda_nid_t nid;
 
 	nid = codec->start_nid;
@@ -384,9 +384,11 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
 		type = get_wcaps_type(get_wcaps(codec, nid));
 		if (type != AC_WID_AUD_IN)
 			continue;
-		*idxp = snd_hda_get_conn_index(codec, nid, pin, false);
-		if (*idxp >= 0)
+		idx = snd_hda_get_conn_index(codec, nid, pin, false);
+		if (idx >= 0) {
+			*idxp = idx;
 			return nid;
+		}
 	}
 	return 0;
 }
@@ -533,7 +535,7 @@ static int add_volume(struct hda_codec *codec, const char *name,
 		      int index, unsigned int pval, int dir,
 		      struct snd_kcontrol **kctlp)
 {
-	char tmp[32];
+	char tmp[44];
 	struct snd_kcontrol_new knew =
 		HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT);
 	knew.private_value = pval;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 502fc94..7696d05 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3348,6 +3348,8 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin,
 
 #define MAX_AUTO_DACS	5
 
+#define DAC_SLAVE_FLAG	0x8000	/* filled dac is a slave */
+
 /* fill analog DAC list from the widget tree */
 static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
 {
@@ -3370,16 +3372,26 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
 /* fill pin_dac_pair list from the pin and dac list */
 static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins,
 			      int num_pins, hda_nid_t *dacs, int *rest,
-			      struct pin_dac_pair *filled, int type)
+			      struct pin_dac_pair *filled, int nums, 
+			      int type)
 {
-	int i, nums;
+	int i, start = nums;
 
-	nums = 0;
-	for (i = 0; i < num_pins; i++) {
+	for (i = 0; i < num_pins; i++, nums++) {
 		filled[nums].pin = pins[i];
 		filled[nums].type = type;
 		filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest);
-		nums++;
+		if (filled[nums].dac) 
+			continue;
+		if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) {
+			filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG;
+			continue;
+		}
+		if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) {
+			filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG;
+			continue;
+		}
+		snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]);
 	}
 	return nums;
 }
@@ -3395,19 +3407,19 @@ static void cx_auto_parse_output(struct hda_codec *codec)
 	rest = fill_cx_auto_dacs(codec, dacs);
 	/* parse all analog output pins */
 	nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs,
-				  dacs, &rest, spec->dac_info,
-				  AUTO_PIN_LINE_OUT);
-	nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
-				  dacs, &rest, spec->dac_info + nums,
-				  AUTO_PIN_HP_OUT);
-	nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
-				  dacs, &rest, spec->dac_info + nums,
-				  AUTO_PIN_SPEAKER_OUT);
+			  dacs, &rest, spec->dac_info, 0,
+			  AUTO_PIN_LINE_OUT);
+	nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
+			  dacs, &rest, spec->dac_info, nums,
+			  AUTO_PIN_HP_OUT);
+	nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
+			  dacs, &rest, spec->dac_info, nums,
+			  AUTO_PIN_SPEAKER_OUT);
 	spec->dac_info_filled = nums;
 	/* fill multiout struct */
 	for (i = 0; i < nums; i++) {
 		hda_nid_t dac = spec->dac_info[i].dac;
-		if (!dac)
+		if (!dac || (dac & DAC_SLAVE_FLAG))
 			continue;
 		switch (spec->dac_info[i].type) {
 		case AUTO_PIN_LINE_OUT:
@@ -3862,7 +3874,7 @@ static void cx_auto_parse_input(struct hda_codec *codec)
 	}
 	if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items)
 		cx_auto_check_auto_mic(codec);
-	if (imux->num_items > 1 && !spec->auto_mic) {
+	if (imux->num_items > 1) {
 		for (i = 1; i < imux->num_items; i++) {
 			if (spec->imux_info[i].adc != spec->imux_info[0].adc) {
 				spec->adc_switching = 1;
@@ -4035,6 +4047,8 @@ static void cx_auto_init_output(struct hda_codec *codec)
 		nid = spec->dac_info[i].dac;
 		if (!nid)
 			nid = spec->multiout.dac_nids[0];
+		else if (nid & DAC_SLAVE_FLAG)
+			nid &= ~DAC_SLAVE_FLAG;
 		select_connection(codec, spec->dac_info[i].pin, nid);
 	}
 	if (spec->auto_mute) {
@@ -4167,9 +4181,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac,
 			     hda_nid_t pin, const char *name, int idx)
 {
 	unsigned int caps;
-	caps = query_amp_caps(codec, dac, HDA_OUTPUT);
-	if (caps & AC_AMPCAP_NUM_STEPS)
-		return cx_auto_add_pb_volume(codec, dac, name, idx);
+	if (dac && !(dac & DAC_SLAVE_FLAG)) {
+		caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+		if (caps & AC_AMPCAP_NUM_STEPS)
+			return cx_auto_add_pb_volume(codec, dac, name, idx);
+	}
 	caps = query_amp_caps(codec, pin, HDA_OUTPUT);
 	if (caps & AC_AMPCAP_NUM_STEPS)
 		return cx_auto_add_pb_volume(codec, pin, name, idx);
@@ -4191,8 +4207,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 	for (i = 0; i < spec->dac_info_filled; i++) {
 		const char *label;
 		int idx, type;
-		if (!spec->dac_info[i].dac)
-			continue;
+		hda_nid_t dac = spec->dac_info[i].dac;
 		type = spec->dac_info[i].type;
 		if (type == AUTO_PIN_LINE_OUT)
 			type = spec->autocfg.line_out_type;
@@ -4211,7 +4226,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 			idx = num_spk++;
 			break;
 		}
-		err = try_add_pb_volume(codec, spec->dac_info[i].dac,
+		err = try_add_pb_volume(codec, dac,
 					spec->dac_info[i].pin,
 					label, idx);
 		if (err < 0)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e125c60..7a73621 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -168,7 +168,7 @@ struct alc_spec {
 	unsigned int auto_mic_valid_imux:1;	/* valid imux for auto-mic */
 	unsigned int automute:1;	/* HP automute enabled */
 	unsigned int detect_line:1;	/* Line-out detection enabled */
-	unsigned int automute_lines:1;	/* automute line-out as well */
+	unsigned int automute_lines:1;	/* automute line-out as well; NOP when automute_hp_lo isn't set */
 	unsigned int automute_hp_lo:1;	/* both HP and LO available */
 
 	/* other flags */
@@ -551,7 +551,7 @@ static void update_speakers(struct hda_codec *codec)
 	if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] ||
 	    spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0])
 		return;
-	if (!spec->automute_lines || !spec->automute)
+	if (!spec->automute || (spec->automute_hp_lo && !spec->automute_lines))
 		on = 0;
 	else
 		on = spec->jack_present;
@@ -565,11 +565,11 @@ static void alc_hp_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (!spec->automute)
-		return;
 	spec->jack_present =
 		detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
 			     spec->autocfg.hp_pins);
+	if (!spec->automute)
+		return;
 	update_speakers(codec);
 }
 
@@ -578,11 +578,15 @@ static void alc_line_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (!spec->automute || !spec->detect_line)
+	/* check LO jack only when it's different from HP */
+	if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0])
 		return;
+
 	spec->line_jack_present =
 		detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
 			     spec->autocfg.line_out_pins);
+	if (!spec->automute || !spec->detect_line)
+		return;
 	update_speakers(codec);
 }
 
@@ -803,7 +807,7 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol,
 	unsigned int val;
 	if (!spec->automute)
 		val = 0;
-	else if (!spec->automute_lines)
+	else if (!spec->automute_hp_lo || !spec->automute_lines)
 		val = 1;
 	else
 		val = 2;
@@ -824,7 +828,8 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol,
 		spec->automute = 0;
 		break;
 	case 1:
-		if (spec->automute && !spec->automute_lines)
+		if (spec->automute &&
+		    (!spec->automute_hp_lo || !spec->automute_lines))
 			return 0;
 		spec->automute = 1;
 		spec->automute_lines = 0;
@@ -1320,7 +1325,9 @@ do_sku:
 	 * 15   : 1 --> enable the function "Mute internal speaker
 	 *	        when the external headphone out jack is plugged"
 	 */
-	if (!spec->autocfg.hp_pins[0]) {
+	if (!spec->autocfg.hp_pins[0] &&
+	    !(spec->autocfg.line_out_pins[0] &&
+	      spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)) {
 		hda_nid_t nid;
 		tmp = (ass >> 11) & 0x3;	/* HP to chassis */
 		if (tmp == 0)
@@ -1784,6 +1791,7 @@ static const char * const alc_slave_vols[] = {
 	"Speaker Playback Volume",
 	"Mono Playback Volume",
 	"Line-Out Playback Volume",
+	"PCM Playback Volume",
 	NULL,
 };
 
@@ -1798,6 +1806,7 @@ static const char * const alc_slave_sws[] = {
 	"Mono Playback Switch",
 	"IEC958 Playback Switch",
 	"Line-Out Playback Switch",
+	"PCM Playback Switch",
 	NULL,
 };
 
@@ -3081,16 +3090,22 @@ static void alc_auto_init_multi_out(struct hda_codec *codec)
 static void alc_auto_init_extra_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	hda_nid_t pin;
+	hda_nid_t pin, dac;
 
 	pin = spec->autocfg.hp_pins[0];
-	if (pin)
-		alc_auto_set_output_and_unmute(codec, pin, PIN_HP,
-						  spec->multiout.hp_nid);
+	if (pin) {
+		dac = spec->multiout.hp_nid;
+		if (!dac)
+			dac = spec->multiout.dac_nids[0];
+		alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac);
+	}
 	pin = spec->autocfg.speaker_pins[0];
-	if (pin)
-		alc_auto_set_output_and_unmute(codec, pin, PIN_OUT,
-					spec->multiout.extra_out_nid[0]);
+	if (pin) {
+		dac = spec->multiout.extra_out_nid[0];
+		if (!dac)
+			dac = spec->multiout.dac_nids[0];
+		alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac);
+	}
 }
 
 /*
@@ -4484,6 +4499,22 @@ static void alc269_fixup_pcm_44k(struct hda_codec *codec,
 	spec->stream_analog_capture = &alc269_44k_pcm_analog_capture;
 }
 
+static void alc269_fixup_stereo_dmic(struct hda_codec *codec,
+				     const struct alc_fixup *fix, int action)
+{
+	int coef;
+
+	if (action != ALC_FIXUP_ACT_INIT)
+		return;
+	/* The digital-mic unit sends PDM (differential signal) instead of
+	 * the standard PCM, thus you can't record a valid mono stream as is.
+	 * Below is a workaround specific to ALC269 to control the dmic
+	 * signal source as mono.
+	 */
+	coef = alc_read_coef_idx(codec, 0x07);
+	alc_write_coef_idx(codec, 0x07, coef | 0x80);
+}
+
 enum {
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -4494,6 +4525,7 @@ enum {
 	ALC275_FIXUP_SONY_HWEQ,
 	ALC271_FIXUP_DMIC,
 	ALC269_FIXUP_PCM_44K,
+	ALC269_FIXUP_STEREO_DMIC,
 };
 
 static const struct alc_fixup alc269_fixups[] = {
@@ -4556,10 +4588,19 @@ static const struct alc_fixup alc269_fixups[] = {
 		.type = ALC_FIXUP_FUNC,
 		.v.func = alc269_fixup_pcm_44k,
 	},
+	[ALC269_FIXUP_STEREO_DMIC] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc269_fixup_stereo_dmic,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
+	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
 	SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index aa376b5..987e3cf 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -673,6 +673,7 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
 static int stac_vrefout_set(struct hda_codec *codec,
 					hda_nid_t nid, unsigned int new_vref)
 {
@@ -696,6 +697,7 @@ static int stac_vrefout_set(struct hda_codec *codec,
 
 	return 1;
 }
+#endif
 
 static unsigned int stac92xx_vref_set(struct hda_codec *codec,
 					hda_nid_t nid, unsigned int new_vref)
@@ -5628,6 +5630,7 @@ again:
 	switch (codec->vendor_id) {
 	case 0x111d76d1:
 	case 0x111d76d9:
+	case 0x111d76df:
 	case 0x111d76e5:
 	case 0x111d7666:
 	case 0x111d7667:
@@ -6571,6 +6574,7 @@ static const struct hda_codec_preset snd_hda_preset_sigmatel[] = {
 	{ .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
+	{ .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 84d8798..4ebfbd8 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -2084,7 +2084,7 @@ static int via_auto_create_speaker_ctls(struct hda_codec *codec)
 	struct via_spec *spec = codec->spec;
 	struct nid_path *path;
 	bool check_dac;
-	hda_nid_t pin, dac;
+	hda_nid_t pin, dac = 0;
 	int err;
 
 	pin = spec->autocfg.speaker_pins[0];
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 6edc67c..493e394 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -1339,6 +1339,10 @@ static u64 hdspm_calc_dds_value(struct hdspm *hdspm, u64 period)
 		break;
 	case MADIface:
 		freq_const = 131072000000000ULL;
+		break;
+	default:
+		snd_BUG();
+		return 0;
 	}
 
 	return div_u64(freq_const, period);
@@ -1356,16 +1360,19 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate)
 
 	switch (hdspm->io_type) {
 	case MADIface:
-	  n = 131072000000000ULL;  /* 125 MHz */
-	  break;
+		n = 131072000000000ULL;  /* 125 MHz */
+		break;
 	case MADI:
 	case AES32:
-	  n = 110069313433624ULL;  /* 105 MHz */
-	  break;
+		n = 110069313433624ULL;  /* 105 MHz */
+		break;
 	case RayDAT:
 	case AIO:
-	  n = 104857600000000ULL;  /* 100 MHz */
-	  break;
+		n = 104857600000000ULL;  /* 100 MHz */
+		break;
+	default:
+		snd_BUG();
+		return;
 	}
 
 	n = div_u64(n, rate);